On Mon, Mar 07, 2011 at 07:29:12AM -0500, Mike Frysinger wrote:
> On Mon, Mar 7, 2011 at 07:15, Mark Brown wrote:
> > It'd seem easier to just merge the two patches together rather than
> > trying to deal with cross-tree issues.
> that sounds like a terrible idea. they're logically different
> changesets and we shouldnt be squashing them together simply to work
> around problems with process workflows.
I'm not saying squash the changes, I'm saying apply them both via the
same tree. From what you're saying the DSP code is used exclusively by
audio devices anyway...
> > Right, and this isn't a particularly unusual requirement especially if
> > the driver isn't going to have any ability to interact with the DSP (at
> > which point it's just coefficient data, the fact that it's a DSP program
> > is uninteresting). The problem is that this isn't a great interface for
> > doing that.
> i dont see suggestions of a better interface anywhere ...
It currently isn't and I'd encourage you to contribute to the discussion
that's been going on on this, or even better help out with code. There
was some discussion on the list recently (in the past month IIRC).
Whatever we do is going to require some additional APIs in alsa-lib, I'd
expect. The key requirements I'm aware of are that we be able to
support:
- Discoverable userspace interface.
- Very large data sizes (megabytes have been mentioned).
- Adding and removing parameter sets at runtime (for in system
callibration).
Given the number of people looking at this from various angles I'd
expect to see some sort of progress this year (probably in the next
quater or so), but obviously no guarantees.
> wrt pm, that's a trivial programming issue that would be resolved in
> the driver. userspace need not care.
That's the ideal, I mentioned this as several of my other review
comments concerned issues with power management in the driver -
addressing those will probably require that the integration occur.
> wrt stream flows, all the customers we've talked to are fine with this
> -- having stream control be an application issue. their application
> is driving the codec directly and knows when it needs to change the
> firmware, so it pauses its alsa flow, loads the new firmware, and
> moves on.
I'd expect that the driver would at least error out if the user tried to
do the wrong thing here, like I say currently the firmware code is just
not joined up with anything else at all.
> that said, i dont see anything in asoc today to address this, so if
> we're simply missing it, please highlight it for us.
There's handling of this in a bunch of drivers for things like EQ
coefficients - the missing bit is a way of telling the driver that there
are new coefficient sets available at runtime, at the minute everything
that supports this enumerates the available coefficient sets in platform
data and presents the user with an enumeration.
> > It's also not an interface which offers any kind of discoverability or
> > enumerability, meaning that it's not suitable for integration into
> > application layer code such as the use case manager. Applications need
> > to be hard coded to know that a particular magic sysfs file needs to be
> > poked. This would be a big step backwards in terms of the ability to
> > run off the shelf application software.
> i dont see the issue here. the firmware is *optional* and does not
> impair basic audio output. further, the firmware is fully
If the firmware is totally optional the driver needs updating -
currently at startup it does:
| + /* Load default program */
| + ret = adau1701_setprogram(codec);
| + if (ret < 0) {
| + printk(KERN_ERR "Loading program data failed\n");
| + goto error;
| + }
which will make the firmware mandatory. In any case, the interface
issues are there if the firmware is optional or not.
> written/compiled/maintained by the end customer, just like the
> application. which means there is no "magic" here -- the end customer
> is the wizard.
> there is no "stock" firmware that ADI or anyone provides for any of
> these SigmaDSP audio codecs.
The question of where the DSP firmware (or coefficient data) comes from
is orthogonal to the issue of runtime management of the data. The issue
is how the application layer can tell if there are multiple coefficient
sets and change between them, either directly or via something like UCM.
Even if the system is using a custom application rather than an off the
shelf distribution (which are becoming more and more common in embedded
systems) there's no reason they shouldn't be able to rely on standard
tools for managing their audio configurations.
At present userspace can enumerate and change the runtime configuration
the system offers via the ALSA APIs (and this will get even better once
the media controller API starts being used). This means that you can
fairly easily write a userspace that'll run on pretty much any Linux
audio hardware, adapting with pure configuration for which you can
provide point and click tuning (realistically by allowing the user to
configure via standard ALSA tools and offering a "save as use case" type
interface). If we start adding backdoors to drivers we're taking a step
back from where we are currently by requiring that the application layer
know magic stuff about individual systems in order to work with them.
If this was an obscure system-specific feature it'd be much less of an
issue but this is something which pretty much all modern CODECs can make
use of.